As a method for encoding a speech signal at low or medium bit rates with high efficiency, there has so far been extensively used a method which separates and encodes the speech signal into linear prediction (LP) coefficients and an excitation signal for driving an LP filter. As typical of such method is code excited linear prediction (CELP). In the CELP, an LP filter, in which the LP coefficients, representing frequency response of the input speech, is driven by an excitation signal represented by the sum of an adaptive codebook (ACB) representing the pitch period of the input speech and a fixed codebook (FCB) composed of random numbers and pulses to generate a synthesized speech signal. The ACB and FCB components are multiplied by gains (ACB gain and FCB gain). As for CELP, reference may be had to M. Schroeder and B. S. Atal: “Code Excited Linear Prediction: High Quality Speech at very low Rates,” Proc. of IEEE Int. Conf. on Acoustics., Speech and Signal Processing, pp. 937 to 940, 1985 (Publication 1).
If the interconnection between a 3G mobile network and a wired packet network is supposed to be implemented, there is raised a problem that direct connection is not possible because of the difference in the standard speech encoding systems used in the respective networks. The simplest solution for this is tandem connection. However, in the tandem connection, speech signals are transiently decoded from a code sequence, obtained on encoding the speech, using one of the standard systems, by this one standard system, and the speech signals, thus decoded, are re-encoded using the other standard system. As a result, there are raised such problems as lowered speech quality, increased delay and increased computation quantity as compared to a case where encoding and decoding are carried out only once in each of the speech encoding/decoding systems.
These problems may effectively be addressed by a transcoding system in which a code obtained on encoding the speech using one of the standard systems into a code decodable using the other standard system in a code domain or in an encoding parameter domain. As for the code converting method, reference may be had to Hong-Goo Kang: “Improving Transcoding Capability of Speech Coders in Clean and Frame Erased Channel Environments,” Proc. of IEEE Workshop on Speech Coding 2000, pp. 78 to 80, 2000 (Publication 2).
FIG. 12 shows an illustrative configuration of a transcoder which converts a code, obtained on encoding the speech using a first speech encoding system (system A) into a code decodable by a second code (system B). Referring to FIG. 12, the transcoder includes an input terminal 10, a code demultiplexing circuit 1010, an LP coefficient code converting circuit 100, an ACB code converting circuit 200, an FCB code converting circuit 300, a gain code converting circuit 400, a code multiplexing circuit 1020 and an output terminal 20. Referring to FIG. 12, the component elements of the conventional transcoder are described.
A first code sequence, obtained on encoding the speech in accordance with the system A, is entered to the input terminal 10.
The code demultiplexing circuit 1010 separates codes corresponding to the LP coefficient, ACB, FCB, ACB gain and FCB gain, that is, LP coefficient code, ACB code, FCB code and the gain code, from the first code sequence, entered to the input terminal 10. The ACB gain and the FCB gain are collectively encoded/decoded and are termed the gains, for simplicity sake. The corresponding codes are termed gain codes. The LP coefficient code, ACB code, FCB code and the gain code are termed first LP coefficient code, first ACB code, first FCB code and the first gain code, respectively. The first LP coefficient codes, first ACB codes, first FCB codes and the first gain code are output to the LP coefficient code converting circuit 100, an ACB code converting circuit 200, an FCB code converting circuit 300 and to the gain code converting circuit 400, respectively.
The LP coefficient code converting circuit 100 is supplied with the first LP coefficient codes, output from the code demultiplexing circuit 1010, to convert the first LP coefficient codes into codes decodable by the system B. The so converted LP coefficient codes are output as the second LP coefficient codes to the code multiplexing circuit 1020.
The ACB code converting circuit 200 is supplied with the first ACB code, output from the code demultiplexing circuit 1010, to convert the first ACB code into a code decodable by the system B. The so converted ACB code is supplied as the second ACB code to the code multiplexing circuit 1020.
The FCB code converting circuit 300 is supplied with the first FCB code, output from the code demultiplexing circuit 1010, to convert the first FCB code into a code decodable by the system B. The so converted FCB code is supplied as the second FCB code to the code multiplexing circuit 1020.
The gain code converting circuit 400 is supplied with the first gain codes, output from the code demultiplexing circuit 1010, to convert the first gain code into code decodable by the system B. The so converted gain code is supplied as the second gain code to the code multiplexing circuit 1020.
More specified operations of the code converting circuits are hereinafter explained
The LP coefficient code converting circuit 100 decodes the first LP coefficient code, entered from the code demultiplexing circuit 1010, by an LP coefficient decoding method in the system A to produce first LP coefficient. The LP coefficient code converting circuit 100 quantizes and encodes the first LP coefficient, in accordance with the quantization method and the encoding method for the LP coefficient by the system B, to yield second LP coefficient code. The LP coefficient code converting circuit 100 outputs the second LP coefficient code to the code multiplexing circuit 1020, as the code decodable by the LP coefficient decoding method by the system B.
The ACB code converting circuit 200 translates the first ACB code, entered from the code demultiplexing circuit 1010, using the relationship of correspondence between the code of the system A and that of the system B, to derive the second ACB code. The ACB code converting circuit 200 outputs the second ACB code to the code multiplexing circuit 1020 as the code decodable by the ACB decoding method in the system B.
The FCB code converting circuit 300 translates the first FCB code, entered from the code demultiplexing circuit 1010, using the relationship of correspondence between the code of the system A and that of the system B, to derive the second FCB code. The FCB code converting circuit 300 outputs the second FCB code to the code multiplexing circuit 1020 as the code decodable by the FCB decoding method in the system B.
The gain code converting circuit 400 decodes the first gain code, supplied from the code demultiplexing circuit 1010, using the gain decoding method of the system A, to produce the first gain. The gain code converting circuit 400 then quantizes and encodes the first gain in accordance with the gain quantization method and the gain encoding method of the system B to derive the second gain and its code (second gain code). The gain code converting circuit 400 then outputs the second gain code as the code decodable by the gain decoding method of the system B to the code multiplexing circuit 1020.
The code multiplexing circuit 1020 is supplied with the second LP coefficient code, output from the LP coefficient code converting circuit 100, the second ACB code, output from the ACB code converting circuit 200, the second FCB code, output from the FCB code converting circuit 300 and the second gain code, output from the gain code converting circuit 400, to output a code sequence, obtained on multiplexing these codes, as a second code sequence to the output terminal 20. The above is the description of FIG. 12.
However, the conventional transcoder, explained above with reference to FIG. 12, suffers from the problem that the sound quality of the background noise energy for the non-speech period is deteriorated.
The reason is that the temporal variation of the background noise energy during the non-speech period are large because of severe temporal changes during the non-speech period of the second gain obtained on re-quantization of the first gain.
Accordingly, it is an object of the present invention to provide a method and an apparatus whereby the deterioration of the sound quality of the background noise during the non-speech period may be reduced, and a recording medium having a corresponding program recorded thereon. Other objects, features and advantages of the present invention will be apparent from the following description.